Many wireless mobile devices are capable of engaging in media services over a wireless network wherein they send and receive media packets. One common such service is a Voice-over-IP (VoIP) call. Network congestion and RF transmission problems can lead to lost packets. Many of the protocols used for media services, like real-time transport protocol (RTP), are focused on packet throughput and efficiency so as to deliver real-time or near-real-time services. Accordingly, lost packets may not be resent or recovered. When packet loss becomes significant, the service quality may become noticeably degraded and the user may have difficulty understanding audio or video. In a VoIP call, packet loss can introduce jitter, clicks, stutter, and other audio artefacts that detract from the user's speech comprehension. Once packet loss becomes significantly high, the media, e.g. audio or video, may become incomprehensible to a user.
It would be advantageous for a mobile device to be able to assess the level of service degradation due to packet loss. Some protocols provide for dissemination of packet statistics. For example, the RTP standard has an associated real-time transport control protocol (RTCP) which provides for the exchange of periodic RTCP packets that contain statistics on the number of packets sent and received. A drawback to this model of gathering packet statistics is that it depends on the sending and receipt of RTCP packets between termination points. If the media path is significantly degraded such that RTP packets are being lost, then there is a reasonable likelihood that an RTCP packet may also be lost.